diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h index be4dcb9c25..1d7f96fa18 100644 --- a/api/peer_connection_interface.h +++ b/api/peer_connection_interface.h @@ -351,7 +351,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface { static const int kUndefined = -1; // Default maximum number of packets in the audio jitter buffer. - static const int kAudioJitterBufferMaxPackets = 50; + static const int kAudioJitterBufferMaxPackets = 200; // ICE connection receiving timeout for aggressive configuration. static const int kAggressiveIceConnectionReceivingTimeout = 1000; diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index 194e8ae365..3f1a5adce3 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h @@ -114,7 +114,7 @@ class AudioReceiveStream { MediaTransportInterface* media_transport = nullptr; // NetEq settings. - size_t jitter_buffer_max_packets = 50; + size_t jitter_buffer_max_packets = 200; bool jitter_buffer_fast_accelerate = false; int jitter_buffer_min_delay_ms = 0; bool jitter_buffer_enable_rtx_handling = false; diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index 1594849cd9..9fc9d08362 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc @@ -272,7 +272,7 @@ void WebRtcVoiceEngine::Init() { options.noise_suppression = true; options.highpass_filter = true; options.stereo_swapping = false; - options.audio_jitter_buffer_max_packets = 50; + options.audio_jitter_buffer_max_packets = 200; options.audio_jitter_buffer_fast_accelerate = false; options.audio_jitter_buffer_min_delay_ms = 0; options.audio_jitter_buffer_enable_rtx_handling = false; diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h index 5b3f864506..3bce78de0a 100644 --- a/media/engine/webrtc_voice_engine.h +++ b/media/engine/webrtc_voice_engine.h @@ -133,7 +133,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface { absl::optional delay_agnostic_aec_; absl::optional experimental_ns_; // Jitter buffer settings for new streams. - size_t audio_jitter_buffer_max_packets_ = 50; + size_t audio_jitter_buffer_max_packets_ = 200; bool audio_jitter_buffer_fast_accelerate_ = false; int audio_jitter_buffer_min_delay_ms_ = 0; bool audio_jitter_buffer_enable_rtx_handling_ = false; diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index ebb2ec28d3..a8935e1f8f 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc @@ -2841,7 +2841,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) { .Times(2) .WillRepeatedly(Return(false)); - EXPECT_EQ(50u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); + EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate); // Nothing set in AudioOptions, so everything should be as default. @@ -2850,7 +2850,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) { EXPECT_TRUE(IsEchoCancellationEnabled()); EXPECT_TRUE(IsHighPassFilterEnabled()); EXPECT_TRUE(IsTypingDetectionEnabled()); - EXPECT_EQ(50u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); + EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate); // Turn typing detection off. diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h index 57fd349fc5..549d355ae0 100644 --- a/modules/audio_coding/neteq/include/neteq.h +++ b/modules/audio_coding/neteq/include/neteq.h @@ -123,7 +123,7 @@ class NetEq { int sample_rate_hz = 16000; // Initial value. Will change with input data. bool enable_post_decode_vad = false; - size_t max_packets_in_buffer = 50; + size_t max_packets_in_buffer = 200; int max_delay_ms = 0; int min_delay_ms = 0; bool enable_fast_accelerate = false;